[Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

Wed Mar 9 00:48:25 MST 2005

First off...  please cancel previous amplification request.  I have  
implemented your ideas with the same errored result.

I am not sure that we are not making it thru authentication.  From my  
digging and comparing packet dumps comparing the soft phone to asterisk  
they have identical transactions through  the ACK reply (the last one  
on the debug below).  The softphone seems to be authenticated after the  
ACK.  I am a newbie to debugging this stuff. I just want to get it  
working.

Thanks everyone in advance for your help.  I am certainly very very  
happy to try anything.

Based on Luki's suggestions I...

Changed sip.conf...

[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=zjh018g8f8
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no

Changed extensions.conf...

exten => _8X.,1, dial(SIP/${EXTEN:1}@broadvoice1,30) ; Dial Broadvoice  
for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;

End result...

Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
<sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3'

SIP debug...

     -- Executing Dial("SIP/6050-132b",  
"SIP/18475098263 at broadvoice1|30") in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:18475098263 at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
To: <sip:18475098263 at sip.broadvoice.com>
Contact: <sip:8475100139 at xxx.xxx.xxx.xxx>
Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205

v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
  (no NAT) to 147.135.8.128:5060
     -- Called 18475098263 at broadvoice1
com*CLI>

Sip read:
INVITE sip:818475098263 at com.imediainc.net SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 <sip:6050 at com.imediainc.net>;tag=7e2776985d5a0891o0
To: <sip:818475098263 at com.imediainc.net>
Call-ID: 26c50864-232ec135 at 64.4.192.110
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest  
username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: 
818475098263 at com.imediainc.net",algorithm=MD5,response="420e39b35648a10c 
129dd4fb5f97ec47"
Contact: 6050 <sip:6050 at 64.4.192.110:5060>
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 <sip:6050 at com.imediainc.net>;tag=7e2776985d5a0891o0
To: <sip:818475098263 at com.imediainc.net>;tag=as2f065f18
Call-ID: 26c50864-232ec135 at 64.4.192.110
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:818475098263 at xxx.xxx.xxx.xxx>
Content-Length: 0

  to 64.4.192.110:5060
com*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
To: <sip:18475098263 at sip.broadvoice.com>
Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE

6 headers, 0 lines
com*CLI>

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
To: <sip:18475098263 at sip.broadvoice.com>;tag=SD38rq699-
Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
CSeq: 102 INVITE
WWW-Authenticate: DIGEST  
realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
Content-Length: 0

8 headers, 0 lines
Transmitting:
ACK sip:18475098263 at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
To: <sip:18475098263 at sip.broadvoice.com>;tag=SD38rq699-
Contact: <sip:8475100139 at xxx.xxx.xxx.xxx>
Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

  (no NAT) to 147.135.8.128:5060
Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
to authenticate on INVITE to '"6050"  
<sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3'

On Mar 9, 2005, at 12:08 AM, Luki wrote:

> Chris,
>
> first of all, if your server has been up for 200 days, I suggest you
> update the kernel -- you don't say if it's Linux, but chances are that
> yes... and there have been some security bugs patched recently.
>
> That aside. I'm not sure, but it's possible that since you are using a
> valid host name ('sip.broadvoice.com') in your dial statement, perhaps
> * tried to talk to it directly and does not consider the section in
> sip.conf. Just a guess. You will notice from the the sip debug output
> that * does not even try to authenticate, as if it didn't know about
> the user/secret.
>
> I use the BV number as the section name, so the dial statement
> essentially looks like: Dial(${EXTEN}@${BV_LINE})
>
> Try changing yours to say "broadvoice" and then the corresponding
> section in sip.conf. I'm using the DCA server, and didn't have an
> issue at all when they introduced INVITE authentication on the
> weekend. This is how my section looks like:
>
> [360350XXXX]
> type=peer
> dtmfmode=inband
> username=360350XXXX
> fromuser=360350XXXX
> secret=XXXXXXXXXX
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> canreinvite=no
> nat=no
> insecure=very
> context=incoming
> outgoinglimit=2
>
> In /etc/hosts I have:
> 147.135.0.128           sip.broadvoice.com
>
> It's the proxy.dca.broadvoice.com server. Hope this helps...
>
> --Luki
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